1. Field of the Invention
The invention relates generally to audio amplification systems, and more particularly to systems and methods for converting data streams at a first sample rate to a second sample rate using a coefficient interpolator that interpolates selectable sets of coefficients.
2. Related Art
Pulse Width Modulation (PWM) or Class D signal amplification technology has existed for a number of years. PWM technology has become more popular with the proliferation of Switched Mode Power Supplies (SMPS). Since this technology emerged, there has been an increased interest in applying PWM techniques in signal amplification applications as a result of the significant efficiency improvement that can be realized through the use of Class D power output topology instead of the legacy (linear Class AB) power output topology.
Early attempts to develop signal amplification applications utilized the same approach to amplification that was being used in the early SMPS. More particularly, these attempts utilized analog modulation schemes that resulted in very low performance applications. These applications were very complex and costly to implement. Consequently, these solutions were not widely accepted. Prior art analog implementations of Class D technology have therefore been unable to displace legacy Class AB amplifiers in mainstream amplifier applications.
Recently, digital PWM modulation schemes have surfaced. These schemes use Sigma-Delta modulation techniques to generate the PWM signals used in the newer digital Class D implementations. These digital PWM schemes, however, did little to offset the major barriers to integration of PWM modulators into the total amplifier solution. Class D technology has therefore continued to be unable to displace legacy Class AB amplifiers in mainstream applications.
One of the problems with conventional implementations of Class D technology lies in the conversion of the digital input data from an input sample rate to an internal sample rate. This process usually involves upsampling and then downsampling the input audio signal to achieve the desired sample rate and filtering the signal either during the up/downsampling or thereafter.
The upsampling and downsampling of the input audio signal may be performed using a polyphase filter. Rather than generating a large number of samples and then throwing away unneeded samples, the polyphase filter generates only those samples that will be retained. In order to achieve good performance, however, it is typically necessary to store 218 filter coefficients that define the filter. It is impractical to store this large number of coefficients. Further, when designing such a filter for this purpose, there are inherent tradeoffs between, for example, passband ripple, stopband rejection and rolloff. Because of the tradeoffs, it is difficult to design a filter that is optimal to cover multiple applications. It would likewise be impractical to have change all of these coefficients to modify the frequency response characteristics for the filter.
A sample rate converter (SRC) based on multirate signal processing requires an interpolation filter. When designing such an FIR low pass filter, there are inherent tradeoffs between passband ripple, stopband rejection and rolloff. This makes it hard to determine an optimal filter to cover multiple applications.